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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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46 MOCK_METHOD2(IncomingRtcpPacket, | 46 MOCK_METHOD2(IncomingRtcpPacket, |
47 int32_t(const uint8_t* incoming_packet, size_t packet_length)); | 47 int32_t(const uint8_t* incoming_packet, size_t packet_length)); |
48 MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)); | 48 MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)); |
49 MOCK_METHOD4(IncomingAudioNTP, | 49 MOCK_METHOD4(IncomingAudioNTP, |
50 int32_t(uint32_t audio_received_ntp_secs, | 50 int32_t(uint32_t audio_received_ntp_secs, |
51 uint32_t audio_received_ntp_frac, | 51 uint32_t audio_received_ntp_frac, |
52 uint32_t audio_rtcp_arrival_time_secs, | 52 uint32_t audio_rtcp_arrival_time_secs, |
53 uint32_t audio_rtcp_arrival_time_frac)); | 53 uint32_t audio_rtcp_arrival_time_frac)); |
54 MOCK_METHOD0(InitSender, int32_t()); | 54 MOCK_METHOD0(InitSender, int32_t()); |
55 MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport)); | 55 MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport)); |
56 MOCK_METHOD1(SetMaxTransferUnit, int32_t(uint16_t size)); | 56 MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)); |
57 MOCK_METHOD3(SetTransportOverhead, | |
58 int32_t(bool tcp, bool ipv6, uint8_t authentication_overhead)); | |
59 MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); | 57 MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); |
60 MOCK_CONST_METHOD0(MaxPayloadLength, uint16_t()); | 58 MOCK_CONST_METHOD0(MaxPayloadSize, size_t()); |
61 MOCK_CONST_METHOD0(MaxDataPayloadLength, uint16_t()); | 59 MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()); |
62 MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec)); | 60 MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec)); |
63 MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec)); | 61 MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec)); |
64 MOCK_METHOD2(RegisterVideoSendPayload, | 62 MOCK_METHOD2(RegisterVideoSendPayload, |
65 void(int payload_type, const char* payload_name)); | 63 void(int payload_type, const char* payload_name)); |
66 MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)); | 64 MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)); |
67 MOCK_METHOD2(RegisterSendRtpHeaderExtension, | 65 MOCK_METHOD2(RegisterSendRtpHeaderExtension, |
68 int32_t(RTPExtensionType type, uint8_t id)); | 66 int32_t(RTPExtensionType type, uint8_t id)); |
69 MOCK_METHOD1(DeregisterSendRtpHeaderExtension, | 67 MOCK_METHOD1(DeregisterSendRtpHeaderExtension, |
70 int32_t(RTPExtensionType type)); | 68 int32_t(RTPExtensionType type)); |
71 MOCK_CONST_METHOD0(StartTimestamp, uint32_t()); | 69 MOCK_CONST_METHOD0(StartTimestamp, uint32_t()); |
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205 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, | 203 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, |
206 StreamDataCountersCallback*(void)); | 204 StreamDataCountersCallback*(void)); |
207 MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&)); | 205 MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&)); |
208 // Members. | 206 // Members. |
209 unsigned int remote_ssrc_; | 207 unsigned int remote_ssrc_; |
210 }; | 208 }; |
211 | 209 |
212 } // namespace webrtc | 210 } // namespace webrtc |
213 | 211 |
214 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ | 212 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ |
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